Re: [asterisk-users] how to used SIPp for sip load testing
On Tue, Dec 27, 2011 at 6:33 AM, virendra bhati virbh...@gmail.com wrote: Hi Sammy, I did the same and start calling. And it's working find but Now I want to the server max capacity with this script then what is the correct process..? There is a nice tutorial on how you can do this in the asterisk source code: ./doc/chan_sip-perf-testing.txt murf On Tue, Dec 27, 2011 at 6:36 PM, Sammy Govind govoi...@gmail.com wrote: Hi, as the Logs say clearly you need to create an extension in default context named service [default] . exten = service,1,NOOP(Incoming call from SIPp) . Regards, Sammy On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.comwrote: Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ m...@parsetree.com ☎ 307-899-5535 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to used SIPp for sip load testing
Hi, as the Logs say clearly you need to create an extension in default context named service [default] . exten = service,1,NOOP(Incoming call from SIPp) . Regards, Sammy On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.com wrote: Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to used SIPp for sip load testing
Hi Sammy, I did the same and start calling. And it's working find but Now I want to the server max capacity with this script then what is the correct process..? On Tue, Dec 27, 2011 at 6:36 PM, Sammy Govind govoi...@gmail.com wrote: Hi, as the Logs say clearly you need to create an extension in default context named service [default] . exten = service,1,NOOP(Incoming call from SIPp) . Regards, Sammy On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.comwrote: Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to used SIPp for sip load testing
Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
